Lattice Vector Quantization Applied to Speech and Audio Coding

Lattice vector quantization （LVQ） has been used for real-time speech and audio coding systems. Compared with conventional vector quantization, LVQ has two main advantages： It has a simple and fast encoding process, and it significantly reduces the amount of memory required. Therefore, LVQ is suitable for use in low-complexity speech and audio coding. In this paper, we describe the basic concepts of LVQ and its advantages over conventional vector quantization. We also describe some LVQ techniques that have been used in speech and audio coding standards of international standards developing organizations （SDOs）.

Bark-Band Residual Noise Model for Parametric Audio Coding

A Bark-band residual noise model integrated with the human hearing mechanism is proposed to efficiently complement sinusoidal model in parametric audio coding. The time-varying spectrum of the residual noise is retrieved by Bark-scale piecewise constant magnitude estimates along with random phases. In the proposed noise model, Bark bands information is obtained by short-time FFT method and window overlap-add technique is exploited to remove boundary discontinuities. SVQ is also incorporated into parameter quantization process for the low bit-rate coding demand. Simulation results and informal listening tests show that when the sinusoidal model is combined with the Bark-band noise model, better synthesis audio quality can be achieved compared with the original sinusoidal modeling audio codec.

Improved Sinusoid Analysis and Post-Processing in Parametric Audio Coding

This paper proposed improvements to the low bit rate parametric audio coder with sinusoid model as its kernel. Firstly, we propose a new method to effectively order and select the perceptually most important sinusoids. The sinusoid which contributes most to the reduction of overall NMR is chosen. Combined with our improved parametric psychoacoustic model and advanced peak riddling techniques, the number of sinusoids required can be greatly reduced and the coding efficiency can be greatly enhanced. A lightweight version is also given to reduce the amount of computation with only little sacrifice of performance. Secondly, we propose two enhancement techniques for sinusoid synthesis: bandwidth enhancement and line enhancement. With little overhead, the effective bandwidth can be extended one more octave; the timbre tends to sound much brighter, thicker and more beautiful.

3D Audio Coding Approach Based on Spatial Perception Features

A new three-dimensional(3D) audio coding approach is presented to improve the spatial perceptual quality of 3D audio. Different from other audio coding approaches, the distance side information is also quantified, and the non-uniform perceptual quantization is proposed based on the spatial perception features of the human auditory system, which is named as concentric spheres spatial quantization(CSSQ) method. Comparison results were presented, which showed that a better distance perceptual quality of 3D audio can be enhanced by 5.7%~8.8% through extracting and coding the distance side information comparing with the directional audio coding, and the bit rate of our coding method is decreased of 8.07% comparing with the spatial squeeze surround audio coding.

Quantization of wavelet packet audio coding

Abstract The method of quantization noise control of audio coding in the wavelet domain is proposed. Using the inverse Discrete Fourier Transform （DFT）, it converts the masking threshold coming from MPEG psycho-acoustic model in the frequency domain to the signal in the time domain; the Discrete Wavelet Packet Transform （DWPF） is performed; the energy in each subband is regarded as the maximum allowed quantization noise energy. The experimental result shows that the proposed method can attain the nearly transparent audio quality below 64kbps for the most testing audio signals.

Review of AVS Audio Coding Standard

Audio Video Coding Standard （AVS） is a second-generation source coding standard and the first standard for audio and video coding in China with independent intellectual property rights. Its performance has reached the international standard. Its coding efficiency is 2 to 3 times greater than that of MPEG -2. This technical solution is more simple, and it can greatly save channel resource. After more than ten years＇ development, AVS has achieved great success. The latest version of the AVS audio coding standard is ongoing and mainly aims at the increasing demand for low bitrate and high quality audio services. The paper reviews the history and recent development of AVS audio coding standard in terms of basic features, key techniques and performance. Finally, the future development of AVS audio coding standard is discussed.

SCALABLE PERCEPTUAL AUDIO REPRESENTATION WITH AN ADAPTIVE THREE TIME-SCALE SINUSOIDAL SIGNAL MODEL

This work is concerned with the development and optimization of a signal model for scalable perceptual audio coding at low bit rates. A complementary two-part signal model consisting of Sines plus Noise (SN) is described. The paper presents essentially a fundamental enhancement to the sinusoidal modeling component. The enhancement involves an audio signal scheme based on carrying out overlap-add sinusoidal modeling at three successive time scales, large, medium, and small. The sinusoidal modeling is done in an analysis-by-synthesis overlap- add manner across the three scales by using a psychoacoustically weighted matching pursuits. The sinusoidal modeling residual at the first scale is passed to the smaller scales to allow for the modeling of various signal features at appropriate resolutions.This approach greatly helps to correct the pre-echo inherent in the sinusoidal model. This improves the perceptual audio quality upon our previous work of sinusoidal modeling while using tile same number of sinusoids. Tile most obvious application for the SN model is in scalable, high fidelity audio coding and signal modification.

High throughput bandwidth optimized VLSI design for motion compensation in AVS HDTV decoder

In this paper we present a motion compensation (MC) design for the newest Audio Video coding Standard (AVS) of China. Because of compression-efficient techniques of variable block size (VBS) and sub-pixel interpolation, intensive pixel calculation and huge memory access are required. We propose a parallel serial filtering mixed luma interpolation data flow and a three-stage multiplication free chroma interpolation scheme. Compared to the conventional designs, the integrated architecture supports about 2.7 times filtering throughput. The proposed MC design utilizes Vertical Z processing order for reference data re-use and saves up to 30% memory bandwidth. The whole design requires 44.3k gates when synthesized at 108 MHz clock frequency using 0.18-μm CMOS technology and can support up to 1920×1088@30 fps AVS HDTV video decoding.

STUDY ON SEMANTIC-BASED VIDEO WATERMARKING METHOD

A new video watermarking method for the Audio Video coding Standard (AVS) is proposed. According to human visual masking properties, this method determines the region of interest for watermark embedding by analyzing video semantics, and generates dynamic robust watermark according to video motion semantics, and embeds watermarks in the Intermediate Frequency (IF) Discrete Cosine Transform (DCT) coefficients of the luminance sub-block prediction residual in the region of interest. This method controls watermark embedding strength adaptively by video textures semantics. Ex- periments show that this method is robust not only to various conventional attacks, but also to re-frame, frame cropping, frame deletion and other video-specific attacks.

Performance Analysis of Data Hiding in MPEG-4 AAC Audio

A high capacity data hiding technique was developed for compressed digital audio. As perceptual audio coding has become the accepted technology for storage and transmission of audio signals, compressed audio information hiding enables robust, imperceptible transmission of data within audio signals, thus allowing valuable information to be attached to the content, such as the song title, lyrics, composer＇s name, and artist or property rights related data. This paper describes simultaneous low bitrate encoding and information hiding for highly compressed audio signals. The information hiding is implemented in the quan- tization process of the audio content which improves robustness, signal quality, and security. The impercep- tibility of the embedded data is ensured based on the masking property of the human auditory system （HAS） The robustness and security are evaluated by various attacking algorithms. Tests with an extended MPEG4 advanced audio coding （AAC） encoder confirm that the method is robust to the regular and singular groups method （RS） and sample pair analysis （SPA） attacks as well as other statistical steganalysis method attacks.

Introduction to AVS Audio

This paper describes a general audio coding algorithm which has been recently standardized by AVS, China. The algorithm is based on a perceptual coding technique. The codec delivers near CD-quality audio at 128kb/s. This paper describes the coder structure in detail and discusses the reasons for specific design methods. A summary of the subjective test results are presented for the prototype codec. Comparison Mean Opinion Score （CMOS） test indicates that the quality of the AVS audio coder is comparable with MPEG Layer-3 audio coder. A reM-time decoder was used for the characterization test, which is based on a 16-bit fixed-point DSP. The performance of the DSP solution was demonstrated, including computational complexity and storage characteristics.

Basic Considerations on AVS DRM Architecture

Digital Rights Management （DRM） is an important infrastructure for the digital media age. It is a part of the AVS （Audio and Video coding Standard） of China. AVS Trusted Decoder （ATD） that plays back digital media program according to rights conditions is the core of AVS DRM architecture. Adaptation layers axe responsible for translating or negotiating between ATD and peripheral systems. The Packaging Adaptation Layer （PAL）, Licensing Adaptation Layer （LAL） and Rendering Adaptation Layer （RAL） will help ATD to gain the interoperability in various DRM environments.